Search Results for "sofia-sip" - Page 3
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Showing 71 open source projects for "sofia-sip"
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VVOIP is voice and video over IP is a P2P application without any protocol like SIP (no server is required) etc.. application. It use decent codecs for video and voice encrypted streams over IP. Moreover it is also comptiable with VoIP modems, you'd con
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Universal Transport For Sip, Including UDP,TCP,HTTP Proxy Tunnelling,Sock4/5,STUN,UPNP,TURN, also a extension in sip for net tunnelling establish.
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SIPatHome was based on the SIP Express Router (SER) which is a well-tested, configurable, free and RFC3261-compliant SIP proxy. The project was providing VoIP server software for the embedded Linux distribution OpenWRT on affordable Linksys WRT hardware.
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Linux port of the Perinatal Information System (SIP) originally developed at CLAP (PAHO / WHO) with extensions. Features a user-friendly GNOME interface and is powered by the PostgreSQL database engine.
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Red Hat Enterprise Linux on Microsoft AzureRed Hat Enterprise Linux (RHEL) on Microsoft Azure provides a secure, reliable, and flexible foundation for your cloud infrastructure. Red Hat Enterprise Linux on Microsoft Azure is ideal for enterprises seeking to enhance their cloud environment with seamless integration, consistent performance, and comprehensive support.
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A SIP/SIMPLE plugin for the Gaim multiprotocol chat client. The project will utilize the GNU oSIP library written in C.
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Distributed SIP Analyzer is a SIP protocol analyzer for Unix. It allows you to examine SIP from different local area network. You can interactively browse the capture data, viewing callflow sequence diagram and detail information for each SIP Session.
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Simple command line IP telephonly application for Linux, that doesn't conform to either H.323 or SIP, doesn't use RTP and is not interoperable with any other program. It does work well though, and has quite good audio quality.
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PNX system develops the middle-ware source code to glue Asterisk with a number of powerful telephony products such as: 1) OpenH323 H.323 stack 2) Vovida SIP stack 3) Bayonne Voice Automation Platform The advantages? An advanced IP PBX supporting the wide
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Bright Data - All in One Platform for Proxies and Web ScrapingBright Data offers the highest quality proxies with automated session management, IP rotation, and advanced web unlocking technology. Enjoy reliable, fast performance with easy integration, a user-friendly dashboard, and enterprise-grade scaling. Powered by ethically-sourced residential IPs for seamless web scraping.
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SEMS is a free (GPL 2) SIP media and application server featuring announcement, voicemail, conference, ivr, B2BUA and more. Please visit our homepage for more.
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SipSpy is a distributed monitoring tool for SIP networks. SpyAgents run on each of the nodes to be monitored, and a SipSpy connects to each of these nodes, receiving information and displaying it in real-time for all the SIP packets monitored.
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This is a SIP based conference server. Allow to host multiple conference bridges and allow multiple participants in each conference.
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The FCP deamon controls the netfilter (linux 2.4) firewall of the localhost by parsing its own protocol. The main purpose of this deamon is to support application level gateways (especially SIP server) to bypass firewall and NAT.
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OpenSnom aims to implement a Fully Free Firmware for the 150Mhz MIPS (INCA-IP - PSB 21553 E) Based SIP Phone Series (320/360/370) The Phone is based on a Embedded Debian with a lot of additional proprietary Software. Until approvement the project is host
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SkypeAsterisk is a full-featured PBX for Skype, SIP, IAX and traditional phone. It is suitable for connecting calls between Skype, PSTN, and IP-PBX. Features include voice mail, conference, call recording and call transferring. An bill system is under de
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This project implements series of homework assignment from Columbia university course. The main target is to create a SIP enabled thin audio client for Linux.
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The SIP2IMS gateway aims to enable simple SIP clients to access IMS networks and provide support for IMS features such as Service-Routes, AKA authentication, IP-Sec/TLS etc.
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Open Source SIP Back to Back User Agent) and Billing system. Has an Embeded TCL interpreter making session control and session mediation limitless in flexibility. Useful for: Routing Engine, Calling Cards, SBC, Peering, Wholesale and more!
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A Simple Middlebox Configuration Protocol (RFC 4540) implementation for Linux. SIMCO is a signaling protocol that can be used by applications (such as SIP B2BUAs) to dynamically control firewalls and Network Address Translators (NATs).
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Tools and libraries for dialing URI addresses directly from telephone's numeric keypad w/o phone directories. Encoded uri (for instance, SIP VoIP phone number) consists of decimal numbers only and can be entered from any telephone or other device.